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audio ADCs with minimum sample rate specs ?

Discussion in 'Electronic Design' started by Winfield Hill, Feb 25, 2006.

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  1. Adam S wrote...
    I'm speculating that there are two possible reasons. First, it's
    common to implement complex CMOS circuits with dynamic rather than
    static circuitry, because it takes fewer MOS transistors. In such
    a case, there will be a minimum operating frequency, although you
    may go below that if you're below the maximum temperature spec.
    The same should go for the charge on node capacitors leaking away.

    Second, some delta sigma modulators have internal integrators that
    have voltages that increase with time, and a maximum clock period
    has to be specified to keep them from overflowing.

    There are lots of cheap high-resolution low-frequency delta-sigma
    ICs available, aimed at scales, geo-electronics, process industry
    and other markets. Why make an awkward use of an audio IC.
     
  2. Al Clark

    Al Clark Guest

    Most sigma delta ADC will actually work at lower sample rates. I am fairly
    certain a Wolfson WM8731 (basically the same as TI 320AIC23 with two pins
    swapped) will work at 4k by just halving the MCLK and using its 8k mode.
    You can ignore the DACs if you don't need them.

    If you are interfacing to a DSP, chances are you have MIPs to burn. You
    could sample at a higher rate and decimate.
     
  3. Adam S

    Adam S Guest

    Why do nearly all audio ADCs and CODECs have a minimum sample rate
    specification ? I'd would like use a low cost 24bit audio ADCs and
    interface to a microcontroller (SPI) at low sample rates < 4kHz.
    Texas Instruments PCM**** range of audio ADCs typically have minimum Fs
    of 16kHz. What happens at lower frequencies ? , does the logic stop
    working, or is it something to do with capacitor charge loss in the
    delta sigma modulator ?
     
  4. Guest

    If they offer a successive approximation ADC with the number of bits
    you need, that might be an answer.

    But perhaps you can use a sigma-delta audio part at it's intended rate,
    but figure out a way to decimate in hardware? For example, if you can
    create the clock with a timer and the L/R signal with a flip flop, and
    then use a timer/interrupt generator to interrupt the micro every n
    clocks and have it record a sample - that leaves you n-1 sample periods
    of time to do other stuff. That does require though that the micro be
    able to grab the serial data at the intended rate.
     
  5. Adam S

    Adam S Guest

    I was attempting to use an 8bit AVR micro, and talk SPI to an audio ADC
    at Fs=4 kHz in order to sample two incoming 1000Hz sine waves at 24bit
    resolution. The software demodulates the two signals at a bandwidth of
    few Hz, before computing the phase and magnitude ratios. The DSP code
    I've written to do the demodulation should in theory allow sample rates
    up to about 16kHz with a 16MHz MPU clock, while consuming 100% processor
    time. So , at much lower sample rates (say 4 to 8kHz) that should leave
    me with plenty of spare clock cycles to play with.

    The problem at those sample rates it seems too low for many audio ADCs
    and too high for many the DC precision ADCs. There doesn't appear to be
    much overlap.

    Adam
     
  6. Guest

    Guest Guest

    : Why do nearly all audio ADCs and CODECs have a minimum sample rate
    : specification ? I'd would like use a low cost 24bit audio ADCs and
    : interface to a microcontroller (SPI) at low sample rates < 4kHz.
    : Texas Instruments PCM**** range of audio ADCs typically have minimum Fs
    : of 16kHz. What happens at lower frequencies ? , does the logic stop
    : working, or is it something to do with capacitor charge loss in the
    : delta sigma modulator ?

    Could be, but more likely, the quantization noise will come up so
    much in the audible band that the output will sound like crap, if you try
    to listen to it. You COULD postfilter the output to remove the
    quantization noise that is in the audio band, but the part will not do
    that for you. Normally, the quantization noise will be guaranteed to be
    out of band (the part is speced within the audio band.)

    I typed that out really quick and I hope that it is clear. Please
    let me know if you need further clarification....

    Joe
     
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